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CN115426590B - A digital audio anti-shake and delay-preventing transmission audio amplifier - Google Patents

A digital audio anti-shake and delay-preventing transmission audio amplifier Download PDF

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CN115426590B
CN115426590B CN202211010443.XA CN202211010443A CN115426590B CN 115426590 B CN115426590 B CN 115426590B CN 202211010443 A CN202211010443 A CN 202211010443A CN 115426590 B CN115426590 B CN 115426590B
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delay
audio
processing module
shake
module
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CN115426590A (en
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杨澄
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Kunshan Haifeiman Technology Group Co ltd
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Guangzhou Gordon Audio Technology Co ltd
Kunshan Haifeiman Technology Group Co ltd
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    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R3/00—Circuits for transducers, loudspeakers or microphones
    • H04R3/02—Circuits for transducers, loudspeakers or microphones for preventing acoustic reaction, i.e. acoustic oscillatory feedback
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R3/00—Circuits for transducers, loudspeakers or microphones
    • H04R3/04—Circuits for transducers, loudspeakers or microphones for correcting frequency response

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  • Engineering & Computer Science (AREA)
  • Acoustics & Sound (AREA)
  • Signal Processing (AREA)
  • Health & Medical Sciences (AREA)
  • General Health & Medical Sciences (AREA)
  • Otolaryngology (AREA)
  • Tone Control, Compression And Expansion, Limiting Amplitude (AREA)

Abstract

本发明涉及一种数字音频防抖及防止延迟传输音频放大器，适合直播、乐器演奏等需要实时反馈和对延时要求极高的应用场景。设置了超频控制器和防抖处理模块，超频控制器获取延时变化曲线，并进行时间积分，将积分结果D输入调节模型，调节模型内计算不同D值对应的超频频率f；超频控制器获得超频频率f并发送至DSP音频处理模块，DSP音频处理模块按照超频频率f工作，从而实现DSP音频处理模块变频工作，降低延迟传输。防抖处理模块用于产生防抖反相信号，并输出至输出接口，防抖反相信号与数模转换放大模块输出的信号叠加，从而实现防抖效果。

The present invention relates to a digital audio anti-shake and delay transmission prevention audio amplifier, which is suitable for application scenarios such as live broadcast and musical instrument performance that require real-time feedback and have extremely high requirements on delay. An overclocking controller and an anti-shake processing module are set up. The overclocking controller obtains the delay change curve and performs time integration. The integral result D is input into the adjustment model, and the overclocking frequency f corresponding to different D values is calculated in the adjustment model; the overclocking controller obtains the overclocking frequency f and sends it to the DSP audio processing module. The DSP audio processing module works according to the overclocking frequency f, thereby realizing the frequency conversion operation of the DSP audio processing module and reducing delayed transmission. The anti-shake processing module is used to generate an anti-shake inverted signal and output it to the output interface. The anti-shake inverted signal is superimposed on the signal output by the digital-to-analog conversion amplifier module, thereby achieving an anti-shake effect.

Description

Digital audio anti-shake and delay transmission preventing audio amplifier
Technical Field
The present invention relates to audio amplifiers, and more particularly, to an audio amplifier for preventing digital audio jitter and delay transmission.
Background
Digital Signal Processing (DSP) is an emerging discipline that involves many disciplines and is widely used in many fields. Since the 60s of the 20 th century, digital signal processing technology has been developed and developed rapidly with the rapid development of computer and information technologies. Digital signal processing has been in widespread use in the field of communications and the like for over twenty years. The digital signal processing is to collect, transform, filter, estimate, enhance, compress, identify, etc. the signal in digital form by using computer or special processing equipment to obtain signal form meeting the needs of people.
The current art is evolving with a new and new trend, and industries such as digital musical instruments, live broadcasting, etc. are evolving rapidly, and in this field, a large number of DSP audio processing devices are required to be used, and at the same time, it is also common to integrate the DSP in the audio amplifier. At present, a DSP module is generally in a fixed frequency working mode, and delay instability is caused when a larger data volume is encountered, so that larger delay and jitter are generated. In order to overcome this situation, the main frequency of DSP devices needs to be continuously increased, but the chip with high main frequency generally has higher cost, and at the same time, the chip can generate larger heat during the operation of high main frequency, which leads to overheating of the chip and reduced service life.
Disclosure of Invention
In order to solve the above problems, the present invention provides a digital audio anti-jitter and delay-transmission-preventing audio amplifier, which comprises an input interface, an output interface, a DSP audio processing module, a digital-to-analog conversion amplifying module, a delay calculating module, a delay feedback module, an over-frequency controller and an anti-jitter processing module;
the input interface is used for connecting with a sound source, acquiring original audio data, and sending the original audio data to the DSP audio processing module;
The DSP audio processing module processes the audio, wherein the processing comprises gain, distortion, EQ adjustment and reverberation increase;
the digital-to-analog conversion amplifying module performs digital-to-analog conversion on the audio data to obtain an analog audio signal, amplifies the analog audio signal and outputs the amplified analog audio signal to the output interface;
The delay calculation module acquires the delay delta t between the input interface and the output interface and sends the delay to the delay feedback module, wherein the delay feedback module sends the delay delta t to the over-frequency controller, and the over-frequency controller is connected with the DSP audio processing module and the anti-shake processing module;
The over-frequency control module is used for controlling the over-frequency performance of the DSP processing chip, and the anti-shake processing module is used for generating an anti-shake reverse signal and outputting the anti-shake reverse signal to the output interface, and the anti-shake reverse signal is overlapped with the signal output by the digital-to-analog conversion amplifying module, so that the anti-shake effect is realized.
The input interface is provided with a marking module which is used for superposing a feature code on the original audio data acquired by the input interface, wherein the feature code comprises a time stamp, and the feature code corresponds to a feature frequency after being digital-to-analog converted;
The feature codes are sent to the DSP audio processing module and then output to the digital-to-analog conversion amplifying module, converted into analog signals and sent to the output interface;
The delay calculating module obtains the characteristic frequency corresponding to the characteristic code from the output interface and calculates the time of the characteristic code, so that the delay delta t of the characteristic code, which is output to the digital-to-analog conversion amplifying module and is transmitted to the output interface after passing through the DSP audio processing module, is calculated.
The delay feedback module acquires the delay delta t obtained by the delay calculation module in real time, generates a time-dependent change curve of the delay delta t, namely a delay change curve, and sends the delay change curve to the over-frequency controller;
The method comprises the steps of obtaining a time delay change curve by an over-frequency controller, carrying out time integration, wherein the time range of integration is from t 1 before the current time to the current time, inputting an integration result D into an adjustment model, calculating over-frequency f corresponding to different D values in the adjustment model, obtaining the over-frequency f by the over-frequency controller and sending the over-frequency f to a DSP audio processing module, and enabling the DSP audio processing module to work according to the over-frequency f, so that the frequency conversion work of the DSP audio processing module is realized, and delay transmission is reduced.
The modeling method of the regulation model comprises the following steps:
Inputting the same section of audio signal into a DSP audio processing module, carrying out single processing and superposition mixing processing by using four processing modes of gain, distortion, EQ adjustment and added reverberation in the DSP audio processing module under the basic working frequency, obtaining 15 groups of delay Deltat 0 under different processing conditions by using a delay calculation module, adjusting the working frequency of the DSP audio processing module to enable the DSP audio processing module to work under different over-frequency f states, and obtaining 15 groups of delay Deltat under different processing conditions under different frequencies by using the delay calculation module to obtain 15 groups of f-Deltat curves;
Setting the delay of the DSP audio processing module in each combination as Deltat 1 when the DSP audio processing module works at the highest super-frequency, calculating a coefficient M= (Deltat 0-△t1)/△t1; the M value represents the sensitivity of the delay of the DSP audio processing module to the working frequency in different working modes, and the larger the M value is, the more sensitive the delay is, and the more obvious the delay reduction is realized by improving the frequency;
The adjustment model is provided with a threshold D 0, when D > D 0, the frequency Deltaf= (k/M). F 0 is increased, and when D < D 0, the frequency Deltaf= (k/M). F 0 is decreased, wherein f 0 is the adjustment step size.
The anti-shake processing module acquires analog audio data from the digital-to-analog conversion amplifying module, and segments the analog audio data by taking the audio loudness A 0 as a threshold value, when the actual loudness A of the analog audio data is larger than A 0, the part with the loudness exceeding A 0 is inverted, when the actual loudness A of the analog audio data is smaller than A 0, the part with the loudness smaller than A 0 is duplicated, and the duplicated loudness is adjusted to be A 0 -A;
the delay feedback module acquires the delay Deltat obtained by the delay calculation module in real time, generates a time-dependent change curve of the delay Deltat, namely a delay change curve, and sends the delay change curve to the anti-shake processing module;
The anti-shake processing module calculates variance of Deltat in a period from the current moment to the current moment from the front t 1, when the variance exceeds a threshold value, the anti-shake processing module outputs the anti-shake audio and the copied audio to the output interface for superposition, and when the variance is smaller than the threshold value, the anti-shake processing module does not output the anti-shake audio and the copied audio to the output interface for superposition.
The DSP audio processing module is further provided with a phase processing module, and the phase processing module is connected between the DSP audio processing module and the digital-to-analog conversion amplifying module and is used for performing phase processing on the audio data processed by the DSP audio processing module.
The phase processing module comprises a relative phase measurer, a phase discriminator, a loop filter, a high-frequency VCO, a synchronous frequency divider and an I2S clock and data phase rearrangement module;
the Data are input into an I2S clock and Data phase rearrangement module, and the I2S clock and Data phase rearrangement module outputs three Data of BCK, WCK and Data after rearrangement;
The audio BCK and the WCK are input to a relative phase measurer and an I2S clock and data phase rearrangement module, meanwhile, one branch of the WCK is input to a phase discriminator, the phase discriminator is output to a loop filter, the loop filter is output to a high-frequency VCO, the high-frequency VCO is output to a synchronous frequency divider and a relative phase measurer, and the synchronous frequency divider is output to the phase discriminator;
the relative phase measurer takes the rising edge of the WCK as a standard, tests the relative phases of the WCK and the BCK, and sends the quantized phase error value to the I2S clock and data phase rearrangement module to realize the relative rearrangement of the clock and data phases.
The rising edge of the BCK clock signal is used as a building reference of the latch signal, the BCK clock signal is aligned to the centers of the rising edge and the falling edge of the Data signal when the phases are arranged, the effective edge of the latch clock is built after the Data is stable, and the Data is ensured not to be influenced by jitter and discrete of circuit delay.
The high frequency VCO is set to 16 times BCK frequency.
The beneficial effects of the invention are as follows:
The invention is provided with an over-frequency controller and an anti-shake processing module, wherein the over-frequency controller acquires a delay change curve, performs time integration, the time range of integration is from t 1 before the current time to the current time, inputs an integration result D into an adjusting model, calculates over-frequency f corresponding to different D values in the adjusting model, acquires the over-frequency f and sends the over-frequency f to a DSP audio processing module, and the DSP audio processing module works according to the over-frequency f, thereby realizing the variable frequency work of the DSP audio processing module, reducing delay transmission, and being particularly suitable for application scenes requiring real-time feedback and extremely high delay requirements such as live broadcasting, musical instrument playing and the like.
The anti-shake processing module is used for generating an anti-shake reverse signal and outputting the anti-shake reverse signal to the output interface, and the anti-shake reverse signal is overlapped with the signal output by the digital-to-analog conversion amplifying module, so that an anti-shake effect is realized. When the delay is unstable, unstable jitter is generated due to the volume of sound played by the loudspeaker caused by the unstable delay, and in order to restrain the jitter, an anti-jitter processing module is used for generating an anti-phase signal or a copy signal when the delay is unstable and is overlapped with the original audio playing, so that the overlarge volume is reduced, the overlarge volume is increased, and the anti-jitter effect is achieved.
Drawings
The accompanying drawings, which are included to provide a further understanding of the disclosed subject matter, are incorporated in and constitute a part of this specification. The drawings also set forth implementations of the disclosed subject matter and, together with the detailed description, serve to explain the principles of the implementations of the disclosed subject matter. No attempt is made to show structural details of the disclosed subject matter in more detail than is necessary for a fundamental understanding of the disclosed subject matter and its various ways of practice.
FIG. 1 is a diagram of the overall architecture of the present invention;
FIG. 2 is a schematic diagram of a phase processing module according to the present invention;
Fig. 3 is an input/output diagram of the phase processing module according to the present invention.
Detailed Description
The advantages, features and manner of attaining the stated objects of the invention will become apparent from the description to follow, and from the drawings.
Example 1:
Referring to fig. 1, a digital audio anti-shake and delay-preventing audio amplifier includes an input interface, an output interface, a DSP audio processing module, a digital-to-analog conversion amplifying module, a delay calculating module, a delay feedback module, an over-frequency controller, and an anti-shake processing module;
the input interface is used for connecting with a sound source, acquiring original audio data, and sending the original audio data to the DSP audio processing module;
The DSP audio processing module processes the audio, wherein the processing comprises gain, distortion, EQ adjustment and reverberation increase;
the digital-to-analog conversion amplifying module performs digital-to-analog conversion on the audio data to obtain an analog audio signal, amplifies the analog audio signal and outputs the amplified analog audio signal to the output interface;
The delay calculation module acquires the delay delta t between the input interface and the output interface and sends the delay to the delay feedback module, wherein the delay feedback module sends the delay delta t to the over-frequency controller, and the over-frequency controller is connected with the DSP audio processing module and the anti-shake processing module;
The over-frequency control module is used for controlling the over-frequency performance of the DSP processing chip, and the anti-shake processing module is used for generating an anti-shake reverse signal and outputting the anti-shake reverse signal to the output interface, and the anti-shake reverse signal is overlapped with the signal output by the digital-to-analog conversion amplifying module, so that the anti-shake effect is realized.
The input interface is provided with a marking module which is used for superposing a feature code on the original audio data acquired by the input interface, wherein the feature code comprises a time stamp, and the feature code corresponds to a feature frequency after being digital-to-analog converted;
The feature codes are sent to the DSP audio processing module and then output to the digital-to-analog conversion amplifying module, converted into analog signals and sent to the output interface;
The delay calculating module obtains the characteristic frequency corresponding to the characteristic code from the output interface and calculates the time of the characteristic code, so that the delay delta t of the characteristic code, which is output to the digital-to-analog conversion amplifying module and is transmitted to the output interface after passing through the DSP audio processing module, is calculated.
The delay feedback module acquires the delay delta t obtained by the delay calculation module in real time, generates a time-dependent change curve of the delay delta t, namely a delay change curve, and sends the delay change curve to the over-frequency controller;
The method comprises the steps of obtaining a time delay change curve by an over-frequency controller, carrying out time integration, wherein the time range of integration is from t 1 before the current time to the current time, inputting an integration result D into an adjustment model, calculating over-frequency f corresponding to different D values in the adjustment model, obtaining the over-frequency f by the over-frequency controller and sending the over-frequency f to a DSP audio processing module, and enabling the DSP audio processing module to work according to the over-frequency f, so that the frequency conversion work of the DSP audio processing module is realized, and delay transmission is reduced.
The modeling method of the regulation model comprises the following steps:
Inputting the same section of audio signal into a DSP audio processing module, carrying out single processing and superposition mixing processing by using four processing modes of gain, distortion, EQ adjustment and added reverberation in the DSP audio processing module under the basic working frequency, obtaining 15 groups of delay Deltat 0 under different processing conditions by using a delay calculation module, adjusting the working frequency of the DSP audio processing module to enable the DSP audio processing module to work under different over-frequency f states, and obtaining 15 groups of delay Deltat under different processing conditions under different frequencies by using the delay calculation module to obtain 15 groups of f-Deltat curves;
Setting the delay of the DSP audio processing module in each combination as Deltat 1 when the DSP audio processing module works at the highest super-frequency, calculating a coefficient M= (Deltat 0-△t1)/△t1; the M value represents the sensitivity of the delay of the DSP audio processing module to the working frequency in different working modes, and the larger the M value is, the more sensitive the delay is, and the more obvious the delay reduction is realized by improving the frequency;
The adjustment model is provided with a threshold D 0, when D > D 0, the frequency Deltaf= (k/M). F 0 is increased, and when D < D 0, the frequency Deltaf= (k/M). F 0 is decreased, wherein f 0 is the adjustment step size.
The anti-shake processing module acquires analog audio data from the digital-to-analog conversion amplifying module, and segments the analog audio data by taking the audio loudness A 0 as a threshold value, when the actual loudness A of the analog audio data is larger than A 0, the part with the loudness exceeding A 0 is inverted, when the actual loudness A of the analog audio data is smaller than A 0, the part with the loudness smaller than A 0 is duplicated, and the duplicated loudness is adjusted to be A 0 -A;
the delay feedback module acquires the delay Deltat obtained by the delay calculation module in real time, generates a time-dependent change curve of the delay Deltat, namely a delay change curve, and sends the delay change curve to the anti-shake processing module;
The anti-shake processing module calculates variance of Deltat in a period from the current moment to the current moment from the front t 1, when the variance exceeds a threshold value, the anti-shake processing module outputs the anti-shake audio and the copied audio to the output interface for superposition, and when the variance is smaller than the threshold value, the anti-shake processing module does not output the anti-shake audio and the copied audio to the output interface for superposition.
Example 2:
The DSP audio processing module is also provided with a phase processing module, and the phase processing module is connected between the DSP audio processing module and the digital-to-analog conversion amplifying module and is used for performing phase processing on the audio data processed by the DSP audio processing module.
Referring to fig. 2-3, the phase processing module includes a relative phase measurer 5, a phase detector 1, a loop filter 2, a high frequency VCO3, a synchronous frequency divider 4, and an I2S clock and data phase rearrangement module 6;
The Data is input into the I2S clock and Data phase rearrangement module 6, and the I2S clock and Data phase rearrangement module 6 outputs three Data of BCK, WCK and Data after rearrangement;
The BCK and the WCK of the audio source are input to a relative phase measurer 5 and an I2S clock and data phase rearrangement module 6, and a branch of the WCK is input to a phase discriminator 1, the phase discriminator 1 is output to a loop filter 2, the loop filter 2 is output to a high-frequency VCO3, the high-frequency VCO3 is output to a synchronous frequency divider 4 and the relative phase measurer 5, and the synchronous frequency divider 4 is output to the phase discriminator 1, so that the phase discriminator 1, the loop filter 2, the high-frequency VCO3 and the synchronous frequency divider 4 form a loop phase-locked frequency synthesizer to generate a high-frequency multiplication clock signal for phase quantization measurement, and the clock is synchronous with the WCK;
The relative phase measurer 5 tests the relative phases of the WCK and the BCK by taking the rising edge of the WCK as a standard, and sends the quantized phase error value to the I2S clock and data phase rearrangement module 6 to realize the relative rearrangement of the clock and data phases.
The rising edge of the BCK clock signal is used as a building reference of the latch signal, the BCK clock signal is aligned to the centers of the rising edge and the falling edge of the Data signal when the phases are arranged, the effective edge of the latch clock is built after the Data is stable, and the Data is ensured not to be influenced by jitter and discrete of circuit delay.
The high frequency VCO3 is set to 16 times BCK frequency.
The above description is merely of the preferred embodiments of the present invention, but the scope of the present invention is not limited thereto, and any person skilled in the art can easily think about the changes or substitutions within the technical scope of the present invention, and the changes or substitutions are intended to be covered by the scope of the present invention. Therefore, the protection scope of the present invention shall be subject to the protection scope of the claims.

Claims (5)

1.一种数字音频防抖及防止延迟传输音频放大器，其特征在于包括输入接口、输出接口、DSP音频处理模块、数模转换放大模块、延时计算模块、延时反馈模块、超频控制器和防抖处理模块；1. A digital audio anti-shake and delay transmission prevention audio amplifier, characterized by comprising an input interface, an output interface, a DSP audio processing module, a digital-to-analog conversion and amplification module, a delay calculation module, a delay feedback module, an overclocking controller and an anti-shake processing module; 输入接口用于连接音源，获取原始音频数据，原始音频数据发送至DSP音频处理模块；The input interface is used to connect the sound source and obtain the original audio data, which is sent to the DSP audio processing module; DSP音频处理模块对音频进行处理，处理包括增益、失真、EQ调节、增加混响；The DSP audio processing module processes the audio, including gain, distortion, EQ adjustment, and adding reverberation; DSP输出的音频数据发送至数模转换放大模块，数模转换放大模块对音频数据进行数模转换，得到模拟音频信号，并将模拟音频信号进行放大后输出至输出接口；The audio data output by the DSP is sent to the digital-to-analog conversion and amplification module, which performs digital-to-analog conversion on the audio data to obtain an analog audio signal, and then amplifies the analog audio signal and outputs it to the output interface; 延时计算模块获取输入接口和输出接口之间的延时△t，并将延时发送至延时反馈模块；延时反馈模块将延时△t发送至超频控制器；超频控制器连接DSP音频处理模块和防抖处理模块；The delay calculation module obtains the delay △t between the input interface and the output interface, and sends the delay to the delay feedback module; the delay feedback module sends the delay △t to the overclocking controller; the overclocking controller is connected to the DSP audio processing module and the anti-shake processing module; 超频控制器用于控制DSP处理芯片的超频性能，防抖处理模块用于产生防抖反相信号，并输出至输出接口，防抖反相信号与数模转换放大模块输出的信号叠加，从而实现防抖效果。The overclocking controller is used to control the overclocking performance of the DSP processing chip. The anti-shake processing module is used to generate an anti-shake inverted signal and output it to the output interface. The anti-shake inverted signal is superimposed on the signal output by the digital-to-analog conversion amplifier module to achieve an anti-shake effect. 2.根据权利要求1所述的数字音频防抖及防止延迟传输音频放大器，其特征在于：2. The digital audio anti-shake and delay-preventing transmission audio amplifier according to claim 1, characterized in that: 输入接口设置有一标记模块，标记模块用于对输入接口获取的原始音频数据叠加一特征码，特征码包括时间戳，同时特征码被数模转换之后对应一特征频率；特征码能够被延时计算模块识别并解析，从而获得特征码的时间戳；The input interface is provided with a marking module, which is used to superimpose a feature code on the original audio data obtained by the input interface, the feature code includes a timestamp, and the feature code corresponds to a feature frequency after being converted from digital to analog; the feature code can be identified and analyzed by the delay calculation module, so as to obtain the timestamp of the feature code; 特征码被发送至DSP音频处理模块后输出至数模转换放大模块，转换成模拟信号发送至输出接口；The feature code is sent to the DSP audio processing module and then output to the digital-to-analog conversion amplifier module, converted into an analog signal and sent to the output interface; 延时计算模块从输出接口处获取特征码对应的特征频率，并计算其时间，从而计算出特征码经过DSP音频处理模块后输出至数模转换放大模块又传输至输出接口的延时△t。The delay calculation module obtains the characteristic frequency corresponding to the characteristic code from the output interface and calculates its time, thereby calculating the delay △t of the characteristic code after passing through the DSP audio processing module, outputting to the digital-to-analog conversion and amplification module, and then transmitting to the output interface. 3.根据权利要求2所述的数字音频防抖及防止延迟传输音频放大器，其特征在于：3. The digital audio anti-shake and delay-preventing transmission audio amplifier according to claim 2, characterized in that: 延时反馈模块实时获取延时计算模块得到的延时△t；并生成延时△t随时间的变化曲线，即延时变化曲线，并将延时变化曲线发送至超频控制器；The delay feedback module obtains the delay △t obtained by the delay calculation module in real time; and generates a curve of the delay △t changing with time, that is, a delay change curve, and sends the delay change curve to the overclocking controller; 超频控制器获取延时变化曲线，并进行时间积分，积分的时间范围为从当前时刻前t1积分至当前时刻；将积分结果D输入调节模型，调节模型内计算不同D值对应的超频频率f；超频控制器获得超频频率f并发送至DSP音频处理模块，DSP音频处理模块按照超频频率f工作，从而实现DSP音频处理模块变频工作，降低延迟传输。The overclocking controller obtains the delay change curve and performs time integration, and the time range of the integration is from t1 before the current moment to the current moment; the integration result D is input into the adjustment model, and the overclocking frequency f corresponding to different D values is calculated in the adjustment model; the overclocking controller obtains the overclocking frequency f and sends it to the DSP audio processing module, and the DSP audio processing module works according to the overclocking frequency f, thereby realizing the frequency conversion operation of the DSP audio processing module and reducing delay transmission. 4.根据权利要求3所述的数字音频防抖及防止延迟传输音频放大器，其特征在于：4. The digital audio anti-shake and delay-preventing transmission audio amplifier according to claim 3, characterized in that: 调节模型的建模方法为：The modeling method of the adjustment model is: 将同一段音频信号输入至DSP音频处理模块，在基础工作频率下，使用DSP音频处理模块中的增益、失真、EQ调节、增加混响四种处理方式进行单一处理和叠加混合处理，共15种组合；使用延时计算模块得到15组不同处理情况下的延时△t0；调节DSP音频处理模块的工作频率，使DSP音频处理模块工作于不同超频频率f状态下，之后使用延时计算模块获得不同频率下的15组不同处理情况下的延时△t，得到15组f-△t曲线；The same audio signal is input into the DSP audio processing module. Under the basic working frequency, four processing methods of gain, distortion, EQ adjustment and reverberation are used in the DSP audio processing module for single processing and superposition mixed processing, with a total of 15 combinations. The delay calculation module is used to obtain 15 groups of delays △ t0 under different processing conditions. The working frequency of the DSP audio processing module is adjusted to make the DSP audio processing module work under different overclocking frequencies f. Then, the delay calculation module is used to obtain 15 groups of delays △t under different processing conditions at different frequencies, and 15 groups of f-△t curves are obtained. 设每一组合中当DSP音频处理模块工作于最高超频频率下时的延时为△t1；计算系数M=(△t0-△t1)/△t1；则M值表征了不同工作模式下DSP音频处理模块的延时对工作频率的敏感度，M值越大，则越敏感，通过提高频率实现的延迟降低越明显；Assume that the delay of the DSP audio processing module in each combination when it works at the highest overclocking frequency is △t 1 ; calculate the coefficient M=(△t 0 -△t 1 )/△t 1 ; then the M value represents the sensitivity of the delay of the DSP audio processing module to the working frequency in different working modes. The larger the M value, the more sensitive it is, and the more obvious the delay reduction achieved by increasing the frequency. 调节模型中设置阈值D0，当D＞D0时，提高频率△f= (k/M)·f0；当D＜D0时，降低频率△f=(k/M)·f0；其中f0为调节步长。A threshold D 0 is set in the adjustment model. When D>D 0 , the frequency △f= (k/M)·f 0 is increased; when D<D 0 , the frequency △f= (k/M)·f 0 is decreased; where f 0 is the adjustment step. 5.根据权利要求2所述的数字音频防抖及防止延迟传输音频放大器，其特征在于5. The digital audio anti-shake and delay-preventing transmission audio amplifier according to claim 2, characterized in that 防抖处理模块从数模转换放大模块获取模拟音频数据，并以音频响度A0为阈值，对模拟音频数据进行分割，当模拟音频数据的实际响度A大于A0时，将响度超过A0的部分反相，当模拟音频数据的实际响度A小于A0时，将响度小于A0的部分进行复制，复制后的响度调整为A0-A；The anti-shake processing module obtains analog audio data from the digital-to- analog conversion and amplification module, and divides the analog audio data with the audio loudness A0 as the threshold. When the actual loudness A of the analog audio data is greater than A0 , the part with the loudness exceeding A0 is inverted. When the actual loudness A of the analog audio data is less than A0 , the part with the loudness less than A0 is copied, and the copied loudness is adjusted to A0 -A. 延时反馈模块实时获取延时计算模块得到的延时△t；并生成延时△t随时间的变化曲线，即延时变化曲线，并将延时变化曲线发送至防抖处理模块；The delay feedback module obtains the delay △t obtained by the delay calculation module in real time; and generates a curve of the delay △t changing with time, that is, a delay change curve, and sends the delay change curve to the anti-shake processing module; 防抖处理模块计算从当前时刻向前t1至当前时刻这一时间段内△t的方差，当方差超过阈值时将防抖处理模块产生的反相音频与复制音频输出至输出接口进行叠加；当方差小于阈值时不将防抖处理模块产生的反相音频与复制音频输出至输出接口进行叠加。The anti-shake processing module calculates the variance of △t in the time period from the current moment to the current moment t 1 forward. When the variance exceeds the threshold, the inverted audio and the copied audio generated by the anti-shake processing module are output to the output interface for superposition; when the variance is less than the threshold, the inverted audio and the copied audio generated by the anti-shake processing module are not output to the output interface for superposition.
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